March 2025 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/1] Swapped l/r channels on ThinkBook 13x Gen4
by MadEnvel 28 May '25

28 May '25

Earlier commits introduced audio support for Lenovo ThinBook 13x Gen 4 that needed a patch due to missing _DSD config. Unfortunately, it looks like the left and right channels in the config are swapped. This patch fixes channel assignment. Tested on Lenovo ThinkBook 13x Gen4. MadEnvel (1): Fix swapped left and right channels for Lenovo ThinBook 13x Gen4 sound sound/pci/hda/cs35l41_hda_property.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) -- 2.45.2

4 4

[PATCH v2] ASoC: amd: use new ACP dev names for DAI links
by Brady Norander 20 May '25

20 May '25

On AMD SoC platforms with an ACP2x gpu ip block (such as stoneyridge), the amdgpu driver will create several platform devices for the ACP ASoC driver to communicate with the ACP hardware block on the gpu. These platform devices include dma for audio and one or multiple i2s interfaces. The amdgpu driver has always created these platform devices with automatic ids. The ASoC machine drives hardcode the platform device name. This creates an issue where if the ACP platform devices are not the first to be created, the ids can be different to what the machine drivers expect, causing them to not find the ACP platform devices and failing to load. Switch to using static ids for these ACP platform devices so that the names never change. Depends on patch: drm/amdgpu: use static ids for ACP platform devs [1] [1] https://lore.kernel.org/all/20250325210517.2097188-1-bradynorander@gmail.co… Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- v2: rewrite commit message to better describe the issue --- sound/soc/amd/acp-da7219-max98357a.c | 8 ++++---- sound/soc/amd/acp-es8336.c | 4 ++-- sound/soc/amd/acp-rt5645.c | 6 +++--- 3 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 02b04f355ca6..42aa009c4e13 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -517,11 +517,11 @@ static const struct snd_soc_ops cz_rt5682_dmic1_cap_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(designware2, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2"))); SND_SOC_DAILINK_DEF(designware3, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.3.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.3"))); SND_SOC_DAILINK_DEF(dlgs, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi"))); @@ -533,7 +533,7 @@ SND_SOC_DAILINK_DEF(adau, DAILINK_COMP_ARRAY(COMP_CODEC("ADAU7002:00", "adau7002-hifi"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link cz_dai_7219_98357[] = { { diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index 0193b3eae7a6..b16dde0e2987 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -137,11 +137,11 @@ static const struct snd_soc_ops st_es8336_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ESSX8336:00", "ES8316 HiFi"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.1.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link st_dai_es8336[] = { { diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index 72ddad24dbda..11d373169380 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -108,15 +108,15 @@ static const struct snd_soc_ops cz_aif1_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(designware2, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2"))); SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link cz_dai_rt5650[] = { { -- 2.49.0

2 2

[PATCH 0/9] ALSA: usb-audio: More fixes / improvements on PCM
by Takashi Iwai 22 Apr '25

22 Apr '25

Hi, this is a patch set for USB-audio to fix and improve the PCM stream handling. The primary target is the enhancement of the low-latency playback that was introduced recently for dealing with the small period size (that is used typically by JACK). There are a few other fixes that are relevant with it. The current patches are found in topic/usb-audio branch of sound.git tree, too. Takashi === Takashi Iwai (9): ALSA: usb-audio: Restrict rates for the shared clocks ALSA: usb-audio: Fix possible race at sync of urb completions ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback ALSA: usb-audio: Disable low-latency playback for free-wheel mode ALSA: usb-audio: Disable low-latency mode for implicit feedback sync ALSA: usb-audio: Check available frames for the next packet size ALSA: usb-audio: Add spinlock to stop_urbs() ALSA: usb-audio: Improved lowlatency playback support ALSA: usb-audio: Avoid killing in-flight URBs during draining sound/usb/card.h | 11 ++- sound/usb/endpoint.c | 230 +++++++++++++++++++++++++++++++------------ sound/usb/endpoint.h | 13 ++- sound/usb/pcm.c | 157 ++++++++++++++++++++++------- 4 files changed, 305 insertions(+), 106 deletions(-) -- 2.26.2

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ALSA 1.2.8 release
by Jaroslav Kysela 15 Apr '25

15 Apr '25

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf tinycompress Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7.2_v1.2.8 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 8

[PATCH v2] ASoC: dwc: always enable/disable i2s irqs
by Brady Norander 08 Apr '25

08 Apr '25

Commit a42e988 ("ASoC: dwc: add DMA handshake control") changed the behavior of the driver to not enable or disable i2s irqs if using DMA. This breaks platforms such as AMD ACP. Audio playback appears to work but no audio can be heard. Revert to the old behavior by always enabling and disabling i2s irqs while keeping DMA handshake control. Fixes: a42e988 ("ASoC: dwc: add DMA handshake control") Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- v2: reword commit message to better describe the issue --- sound/soc/dwc/dwc-i2s.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 57b789d7fbed..5b4f20dbf7bb 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -199,12 +199,10 @@ static void i2s_start(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 1); - /* I2S needs to enable IRQ to make a handshake with DMAC on the JH7110 SoC */ - if (dev->use_pio || dev->is_jh7110) - i2s_enable_irqs(dev, substream->stream, config->chan_nr); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_enable_dma(dev, substream->stream); + i2s_enable_irqs(dev, substream->stream, config->chan_nr); i2s_write_reg(dev->i2s_base, CER, 1); } @@ -218,11 +216,12 @@ static void i2s_stop(struct dw_i2s_dev *dev, else i2s_write_reg(dev->i2s_base, IRER, 0); - if (dev->use_pio || dev->is_jh7110) - i2s_disable_irqs(dev, substream->stream, 8); - else + if (!(dev->use_pio || dev->is_jh7110)) i2s_disable_dma(dev, substream->stream); + i2s_disable_irqs(dev, substream->stream, 8); + + if (!dev->active) { i2s_write_reg(dev->i2s_base, CER, 0); i2s_write_reg(dev->i2s_base, IER, 0); -- 2.49.0

2 1

[PATCH alsa-ucm-conf 1/2] sof-soundwire: cs42l43: Drop headset mic from mic mute LED
by Maciej Strozek 01 Apr '25

01 Apr '25

As per conversation in [1], let headsets be dropped from LED bindings. Also remove turning off the mics on device disable since they are now not needed. [1]: https://github.com/alsa-project/alsa-ucm-conf/pull/512 Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- ucm2/codecs/cs42l43/init.conf | 6 ------ ucm2/sof-soundwire/cs42l43-dmic.conf | 2 -- ucm2/sof-soundwire/cs42l43.conf | 1 - 3 files changed, 9 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 0e987c21..c0fa863c 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -10,9 +10,3 @@ LibraryConfig.remap.Config { } } } - -BootSequence [ - cset "name='cs42l43 Decimator 1 Switch' 0" -] - -Macro [{ SetLED { LED="mic" Action="attach" CtlId="cs42l43 Decimator 1 Switch" } }] diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index 70ce40ff..b68a70d1 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -13,8 +13,6 @@ SectionDevice."Mic" { ] DisableSequence [ - cset "name='cs42l43 Decimator 3 Switch' 0" - cset "name='cs42l43 Decimator 4 Switch' 0" cset "name='cs42l43 DP1TX1 Input' 'None'" cset "name='cs42l43 DP1TX2 Input' 'None'" ] diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 4b4e1187..51832126 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -33,7 +33,6 @@ SectionDevice."Headset" { ] DisableSequence [ - cset "name='cs42l43 Decimator 1 Switch' 0" cset "name='cs42l43 DP2TX1 Input' 'None'" cset "name='cs42l43 DP2TX2 Input' 'None'" ] -- 2.39.5

2 6

Difference in behaviour between physical devices and virtual endpoints
by GitHub issues - opened 31 Mar '25

31 Mar '25

alsa-project/alsa-lib issue #447 was opened from reuk: As a disclaimer, there's a good chance this is user error/misunderstanding rather than a bug, but I'd appreciate any insights to help me understand the behaviour I'm seeing. # Environment - Fedora 41 x86_64 - alsa-lib-devel-1.2.13-3 # Steps - Run `aseqdump -u 1` in a terminal - Connect ProtoZOA hardware via USB - Launch QjackCtl -> Graph, then connect the ProtoZOA's MIDI 2.0 output port (0) to `aseqdump`'s input port. <img width="535" alt="Image" src="https://github.com/user-attachments/assets/17bf6a74-f83d-41b4-afc6-05413a7a…" /> - Observe that `aseqdump` reports a new subscription from `24:0 -> 128:0` (the client IDs may vary, the important thing is that the subscription is between to ports with ID `0`). - Move one of the controllers on the ProtoZOA. Observe that `aseqdump` reports channel voice events on group 0 with source `24:1`. <img width="502" alt="Image" src="https://github.com/user-attachments/assets/2dfc5929-123f-4e33-abbb-ea0c2635…" /> - Connect `aseqdump` to port `1:Group 1 (FB1, FB2)`, and interact with the ProtoZOA. Observe that `aseqdump` still reports channel voice events on group 0 with source `24:1`. - Now, build and run the [seq-ump-example](https://github.com/alsa-project/alsa-lib/blob/master/test/…. - Connect the ProtoZOA to the ACMESynth input, and the ACMESynth output to `aseqdump` as shown. Observe that `aseqdump` does **not** report channel voice events when interacting with the ProtoZOA. <img width="790" alt="Image" src="https://github.com/user-attachments/assets/71349cd6-14b8-4df9-8ee6-ca1091c2…" /> - Finally, update the connections as shown, so that `aseqdump` is connected to ACMESynth's `1:Group 1 (Filter 1)` output. Observe that `aseqdump` now reports channel voice events with source `128:1` when interacting with the ProtoZOA. <img width="772" alt="Image" src="https://github.com/user-attachments/assets/57f4e767-b2d6-4156-8514-578851d5…" /> # TL;DR When we subscribe to a USB `MIDI 2.0` port, we receive messages from all of the endpoint's ports. Messages are dispatched to the appropriate port based on the message group. When we subscribe to a virtual UMP endpoint's `MIDI 2.0` port, this appears to *only* subscribe to that single port, and not to the rest of the ports on the endpoint, so that messages addressed to specific groups get lost. # Questions - Is this difference in behaviour between hardware and virtual endpoints intentional? - What is the "correct" way to subscribe to all UMP events from an endpoint? Should I subscribe just to the `MIDI 2.0` endpoint, or to all ports on the endpoint, or something else? Issue URL : https://github.com/alsa-project/alsa-lib/issues/447 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH alsa-ucm-conf v1] sof-soundwire: Add LED support for cs35l56 amplifiers
by Stefan Binding 31 Mar '25

31 Mar '25

Systems may have up to 8 CS35L56 amplifiers, and each amplifier should be attached to the speaker LED. A mapped Playback switch is required to ensure all amps are muted and unmute appropriately. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- ucm2/codecs/cs35l56-bridge/init.conf | 11 ++++++++ ucm2/codecs/cs35l56/init.conf | 36 +++++++++++++++++++++++++++ ucm2/sof-soundwire/cs35l56.conf | 32 ++---------------------- ucm2/sof-soundwire/sof-soundwire.conf | 4 +-- 4 files changed, 51 insertions(+), 32 deletions(-) create mode 100644 ucm2/codecs/cs35l56-bridge/init.conf create mode 100644 ucm2/codecs/cs35l56/init.conf diff --git a/ucm2/codecs/cs35l56-bridge/init.conf b/ucm2/codecs/cs35l56-bridge/init.conf new file mode 100644 index 000000000000..407bb0937559 --- /dev/null +++ b/ucm2/codecs/cs35l56-bridge/init.conf @@ -0,0 +1,11 @@ +# cs35l56 specific control settings + +FixedBootSequence [ + cset "name='AMPL Speaker Switch' 0" + cset "name='AMPR Speaker Switch' 0" +] + +Macro [ + { SetLED { LED="speaker" Action="attach" CtlId="AMPL Speaker Switch" } } + { SetLED { LED="speaker" Action="attach" CtlId="AMPR Speaker Switch" } } +] \ No newline at end of file diff --git a/ucm2/codecs/cs35l56/init.conf b/ucm2/codecs/cs35l56/init.conf new file mode 100644 index 000000000000..dca14c709fda --- /dev/null +++ b/ucm2/codecs/cs35l56/init.conf @@ -0,0 +1,36 @@ +# cs35l56 specific control settings + +# +# Arguments: +# ForAmps - regex filter for var:SpeakerAmps +# Amp - amplifier number 1-8 +# +DefineMacro.cs35l56spkInit.If.0 { + Condition { + Type RegexMatch + Regex "${var:__ForAmps}" + String "${var:SpeakerAmps}" + } + True { + LibraryConfig.remap.Config.ctl.default.map { + "name='cs35l56 Speaker Playback Switch'" { + "name='AMP${var:__Amp} Speaker Switch'".vindex."${eval:($__Amp-1)}" 0 + } + } + FixedBootSequence [ + cset "name='AMP${var:__Amp} Speaker Switch' 0" + ] + Macro [{ SetLED { LED="speaker" Action="attach" CtlId="AMP${var:__Amp} Speaker Switch" } }] + } +} + +Macro [ + { cs35l56spkInit { ForAmps "[1-8]" Amp 1 } } + { cs35l56spkInit { ForAmps "[2-8]" Amp 2 } } + { cs35l56spkInit { ForAmps "[3-8]" Amp 3 } } + { cs35l56spkInit { ForAmps "[4-8]" Amp 4 } } + { cs35l56spkInit { ForAmps "[5-8]" Amp 5 } } + { cs35l56spkInit { ForAmps "[6-8]" Amp 6 } } + { cs35l56spkInit { ForAmps "[78]" Amp 7 } } + { cs35l56spkInit { ForAmps "8" Amp 8 } } +] diff --git a/ucm2/sof-soundwire/cs35l56.conf b/ucm2/sof-soundwire/cs35l56.conf index f74b99afbb55..53e4d6dd8eda 100644 --- a/ucm2/sof-soundwire/cs35l56.conf +++ b/ucm2/sof-soundwire/cs35l56.conf @@ -1,40 +1,12 @@ # Use case Configuration for sof-soundwire card -# -# Arguments: -# ForAmps - regex filter for var:SpeakerAmps -# Amp - amplifier number 1-8 -# -DefineMacro.cs42l43spk.If.0 { - Condition { - Type RegexMatch - Regex "${var:__ForAmps}" - String "${var:SpeakerAmps}" - } - True { - EnableSequence [ - cset "name='AMP${var:__Amp} Speaker Switch' 1" - ] - DisableSequence [ - cset "name='AMP${var:__Amp} Speaker Switch' 0" - ] - } -} - SectionDevice."Speaker" { Comment "Speaker" - Macro.num1.cs42l43spk { ForAmps "[1-8]" Amp 1 } - Macro.num2.cs42l43spk { ForAmps "[2-8]" Amp 2 } - Macro.num3.cs42l43spk { ForAmps "[3-8]" Amp 3 } - Macro.num4.cs42l43spk { ForAmps "[4-8]" Amp 4 } - Macro.num5.cs42l43spk { ForAmps "[5-8]" Amp 5 } - Macro.num6.cs42l43spk { ForAmps "[6-8]" Amp 6 } - Macro.num7.cs42l43spk { ForAmps "[78]" Amp 7 } - Macro.num8.cs42l43spk { ForAmps "8" Amp 8 } - Value { PlaybackPriority 100 PlaybackPCM "hw:${CardId},2" + PlaybackMixer "default:${CardId}" + PlaybackMixerElem "cs35l56 Speaker" } } diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 1cddfbf4cfbf..8f9bc82c2d3b 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -102,10 +102,10 @@ If.multi_headset { } } -If.spk_init_rt1318 { +If.spk_init { Condition { Type RegexMatch - Regex "(rt1318(-1)?)" + Regex "(rt1318(-1)?|cs35l56(-bridge)?)" String "${var:SpeakerCodec1}" } True.Include.spk_init.File "/codecs/${var:SpeakerCodec1}/init.conf" -- 2.43.0

2 1

USB-Audio: Add focusrite scarlett 18i20 lineup
by GitHub pull_request - opened 30 Mar '25

30 Mar '25

alsa-project/alsa-ucm-conf pull request #539 was opened from ScarletRav3n: Other solutions exist to add full mixing and routing with these devices. This PR adds basic functionality in an out-of-the-box style for 1st gen, 2nd gen, 3rd gen, and 4th gen. IO includes XLR, Line, S/PDIF, ADAT, and Headphone ports. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/539 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/539.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add Focusrite Scarlett 18i20 Lineup
by GitHub pull_request - edited 30 Mar '25

30 Mar '25

alsa-project/alsa-ucm-conf pull request #538 was edited from ScarletRav3n: Other solutions exist to add full mixing and routing with these devices. This PR adds basic functionality in an out-of-the-box style for 1st gen, 2nd gen, 3rd gen, and 4th gen. I/O includes all XLR, Line, S/PDIF, ADAT, and Headphone ports. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add Focusrite Scarlett 18i20 Lineup
by GitHub pull_request - opened 30 Mar '25

30 Mar '25

alsa-project/alsa-ucm-conf pull request #538 was opened from ScarletRav3n: Other solutions exist to add full mixing and routing with these devices. This PR adds basic functionality in an out-of-the-box style for 2nd gen, 3rd gen, and 4th gen. 1st gen is not supported with this iteration of PR. I/O includes all XLR, Line, S/PDIF, ADAT, and Headphone ports. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

envy24control: fix memory access errors in profiles
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #34 was opened from andreas56: Fix two memory errors in the profiles parser: an invalid read and a "source and destination overlap in strncpy" warning. When the profiles page is initialized it fetches the profile names from the profiles file. When a profile wasn't defined in the file, the parser made invalid reads outside the buffer. Request URL : https://github.com/alsa-project/alsa-tools/pull/34 Patch URL : https://github.com/alsa-project/alsa-tools/pull/34.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix file descriptor leaks in profiles
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #33 was opened from andreas56: File descriptors were leaked when "Save active profile" was pressed. Add the missing calls to close. Request URL : https://github.com/alsa-project/alsa-tools/pull/33 Patch URL : https://github.com/alsa-project/alsa-tools/pull/33.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix crash when using system profiles file
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #32 was opened from andreas56: envy24control crashed if you tried to start it with -f /etc/envy24control/profiles.conf. Request URL : https://github.com/alsa-project/alsa-tools/pull/32 Patch URL : https://github.com/alsa-project/alsa-tools/pull/32.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix GtkSpinButton runtime warnings
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #31 was opened from andreas56: Change the parameters to gtk_adjustment_new to get rid of this runtime warning from GTK: GtkSpinButton: setting an adjustment with non-zero page size is deprecated The change is also important when envy24control is ported to GTK 3, as the spin buttons will stop working if it's not done. Request URL : https://github.com/alsa-project/alsa-tools/pull/31 Patch URL : https://github.com/alsa-project/alsa-tools/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

envy24control: fix control callback for spdif output
by GitHub pull_request - opened 29 Mar '25

29 Mar '25

alsa-project/alsa-tools pull request #30 was opened from andreas56: Update S/PDIF Output Settings GUI when settings change. Request URL : https://github.com/alsa-project/alsa-tools/pull/30 Patch URL : https://github.com/alsa-project/alsa-tools/pull/30.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

[PATCH alsa-ucm-conf v2 1/2] sof-soundwire: cs42l43: Drop headset mic from mic mute LED
by Maciej Strozek 28 Mar '25

28 Mar '25

As per conversation in [1], let headsets be dropped from LED bindings. [1]: https://github.com/alsa-project/alsa-ucm-conf/pull/512 Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v2: Added DisableSequences back --- ucm2/codecs/cs42l43/init.conf | 6 ------ 1 file changed, 6 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 0e987c21..c0fa863c 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -10,9 +10,3 @@ LibraryConfig.remap.Config { } } } - -BootSequence [ - cset "name='cs42l43 Decimator 1 Switch' 0" -] - -Macro [{ SetLED { LED="mic" Action="attach" CtlId="cs42l43 Decimator 1 Switch" } }] -- 2.39.5

1 1

[PATCH v1] ALSA: hda/tas2781: Upgrade calibratd-data writing code to support Alpha and Beta dsp firmware
by Shenghao Ding 28 Mar '25

28 Mar '25

Since 2025, the firmware for tas2781 has been added more audio acoustic features, such as non-linear compensation, advanced battery guard, rattle-noise suppression, etc. The version was divided into two different series. Both series have a slight change on the calibrated data storage addresses, which becames flexible instead of fixed. In order to support new firwmares in time, the code have some related upgrades. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - revise the subject and description. v1: - Add updating calibration addresses code into tas2781_apply_calib in case of Alpha and Beta firmware. --- sound/pci/hda/tas2781_hda_i2c.c | 30 ++++++++++++++++++++---------- 1 file changed, 20 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 9ed49b0dbe6b..29dc4f500580 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -558,28 +558,38 @@ static int tas2563_save_calibration(struct tasdevice_priv *tas_priv) static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) { - static const unsigned char page_array[CALIB_MAX] = { - 0x17, 0x18, 0x18, 0x13, 0x18, + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + unsigned int cali_reg[CALIB_MAX] = { + TASDEVICE_REG(0, 0x17, 0x74), + TASDEVICE_REG(0, 0x18, 0x0c), + TASDEVICE_REG(0, 0x18, 0x14), + TASDEVICE_REG(0, 0x13, 0x70), + TASDEVICE_REG(0, 0x18, 0x7c), }; - static const unsigned char rgno_array[CALIB_MAX] = { - 0x74, 0x0c, 0x14, 0x70, 0x7c, - }; - int offset = 0; int i, j, rc; + int oft = 0; __be32 data; + if (tas_priv->dspbin_typ != TASDEV_BASIC) { + cali_reg[0] = r->r0_reg; + cali_reg[1] = r->invr0_reg; + cali_reg[2] = r->r0_low_reg; + cali_reg[3] = r->pow_reg; + cali_reg[4] = r->tlimit_reg; + } + for (i = 0; i < tas_priv->ndev; i++) { for (j = 0; j < CALIB_MAX; j++) { data = cpu_to_be32( - *(uint32_t *)&tas_priv->cali_data.data[offset]); + *(uint32_t *)&tas_priv->cali_data.data[oft]); rc = tasdevice_dev_bulk_write(tas_priv, i, - TASDEVICE_REG(0, page_array[j], rgno_array[j]), - (unsigned char *)&data, 4); + cali_reg[j], (unsigned char *)&data, 4); if (rc < 0) dev_err(tas_priv->dev, "chn %d calib %d bulk_wr err = %d\n", i, j, rc); - offset += 4; + oft += 4; } } } -- 2.34.1

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[ alsa-lib ] OpenHarmony Adaptation Proposal
by GitHub issues - opened 28 Mar '25

28 Mar '25

alsa-project/alsa-lib issue #445 was opened from Asercn: # [ alsa-lib ] OpenHarmony Adaptation Proposal ## Background OpenHarmony is an open-source project incubated and operated by the OpenAtom Open Source Foundation. Its goal is to build an operating system framework and platform for intelligent terminal devices in the era of full scenarios, full connectivity, and full intelligence. ALSA, as an important audio architecture in Linux systems, allows developers to use Linux system calls in programs to control and manage the underlying operating system. Due to its efficient and easy-to-use features, it is also suitable for embedded devices, which aligns well with OpenHarmony's concept of supporting a variety of devices. Currently, ALSA is widely used in the mobile application development field of OpenHarmony. We will submit the adaptation code via a pull request (PR) as soon as possible. This issue is created to engage in in-depth discussions with the community about integrating the adaptation code. We sincerely welcome any valuable suggestions. ## Adaptation Plan 1. Details of the ALSA adaptation plan: 2. Submit back to the main community. 3. Ongoing community maintenance. ## Test Results We have completed large-scale compatibility testing, covering different versions of HarmonyOS and various types of HarmonyOS devices. The test results show that the modified version is highly compatible with the HarmonyOS system and no significant stability issues have been observed. Benefits to the Upstream Community * Can reach OpenHarmony’s large user base and developer community. * By supporting the development trends of the operating system, it ensures the long-term development potential of the project. * Enhances the project's processing capability in diverse operating environments. Issue URL : https://github.com/alsa-project/alsa-lib/issues/445 Repository URL: https://github.com/alsa-project/alsa-lib

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Add UCM2 support for ASUS ROG Strix B850-I Gaming WiFi
by GitHub pull_request - edited 28 Mar '25

28 Mar '25

alsa-project/alsa-ucm-conf pull request #537 was edited from austinthiel: [Link to open issue](https://github.com/alsa-project/alsa-ucm-conf/issues/534#issue-29483… I referenced [this guide](https://askubuntu.com/a/1469066) to get SPDIF out working on my ASUS B850-I Gaming WiFi [alsa-info.txt](https://github.com/user-attachments/files/19499096/alsa-info… In case it's relevant, using a sample rate of 48kHz is required. I found that other sample rate produces very low quality, garbled audio. I read that this is because SPDIF only supports 48kHz in hardware, and any other sample rates require the Realtek drivers. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Add UCM2 support for ASUS ROG Strix B850-I Gaming WiFi
by GitHub pull_request - edited 28 Mar '25

28 Mar '25

alsa-project/alsa-ucm-conf pull request #537 was edited from austinthiel: Referenced [this guide](https://askubuntu.com/a/1469066) to get SPDIF out working on my ASUS B850-I Gaming WiFi [alsa-info.txt](https://github.com/user-attachments/files/19499096/alsa-info… In case it's relevant, using a sample rate of 48kHz is required. I found that other sample rate produces very low quality, garbled audio. I read that this is because SPDIF only supports 48kHz in hardware, and any other sample rates require the Realtek drivers. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM2 support for ASUS ROG Strix B850-I Gaming WiFi
by GitHub pull_request - edited 28 Mar '25

28 Mar '25

alsa-project/alsa-ucm-conf pull request #537 was edited from austinthiel: Referenced [this guide](https://askubuntu.com/a/1469066) to get SPDIF out working on my ASUS B850-I Gaming WiFi [alsa-info.txt](https://github.com/user-attachments/files/19499096/alsa-info… In case it's relevant, using a sample rate of 48kHz is required. I found that other sample rate produces very low quality, garbled audio. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCM2 support for ASUS ROG Strix B850-I Gaming WiFi
by GitHub pull_request - opened 28 Mar '25

28 Mar '25

alsa-project/alsa-ucm-conf pull request #537 was opened from austinthiel: Referenced [this guide](https://askubuntu.com/a/1469066) to get SPDIF out working on my ASUS B850-I Gaming WiFi [alsa-info.txt](https://github.com/user-attachments/files/19498803/alsa-info… In case it's relevant, using a sample rate of 48kHz is required. I found that other sample rate produces very low quality, garbled audio. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/537.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio Add Roland BridgeCast One USB-Audio.conf
by GitHub pull_request - opened 27 Mar '25

27 Mar '25

alsa-project/alsa-ucm-conf pull request #536 was opened from Simon99de: Add Roland BridgeCast One uses the same config as the bridge cast v2 as it has the same outputs and it worked in my test. 0582:030d Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/536 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/536.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: ALC4080 - add rear microphone support for 0414:a014
by GitHub pull_request - opened 26 Mar '25

26 Mar '25

alsa-project/alsa-ucm-conf pull request #535 was opened from perexg: 0414:a014 Gigabyte B850I Aorus Pro (rev 1.0) Link: https://github.com/alsa-project/alsa-ucm-conf/issues/528 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/535 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/535.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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